This invention relates to an adaptive digital filter capable of generating an arbitrary transfer function. In particular, it relates to an adaptive digital filter having a fast rate of convergence, suitable for use in a device such as an echo canceler. The invention also relates to an echo canceler incorporating such an adaptive digital filter.
Recent rapid progress in digital signal-processing technology has created great interest in adaptive digital filters due to their wide range of applications. Typical of these applications is system identification, which is a process of estimating an unknown system characteristic from input and output data.
Means for the identification of an unknown system by use of an adaptive digital filter are shown in schematic form in FIG. 1. These means comprise a signal input terminal 11, an error output terminal 12, an unknown system 13, an adaptive digital filter (ADF) 14, and an adder 15. In the figure, x(k) is the input to the unknown system 13 and the adaptive digital filter 14 at time k, y(k) is the output from the unknown system 13 at time k, y(k) is the output from the adaptive digital filter 14 at time k, e(k) is the estimation error at time k, H(z) is the transfer function of the unknown system, and H(z) is the transfer function of the adaptive digital filter 14. In the configuration shown, if the evaluation function is J=e(k).sup.2, then when J=0 the adaptive digital filter 14 is regarded as correctly estimating the characteristic of the unknown system 13.
A specific type of device using an adaptive digital filter like the one described above is an echo canceler. Echo cancelers are used, for example, in teleconferencing systems, for which there has been a recently growing demand. FIG. 2 is a schematic diagram of a teleconferencing system employing an echo canceler. This system comprises a pair of microphones 21-1 and 21-2, a pair of loudspeakers 22-1 and 22-2, a pair of echo cancelers 23-1 and 23-2, which respectively include adaptive digital filters 25-1 and 25-2, and a pair of transmission lines 24-1 and 24-2, and has a pair of acoustically coupled paths 26-1 and 26-2, In most teleconferencing systems the loudspeaker and microphone shown in FIG. 2 are integrated into a single unit called a voice terminal. This gives rise to an acoustic coupling between the loudspeaker and the microphone: the signal output from the loudspeaker is coupled into the microphone and greatly degrades the quality of the voice transmission. In FIG. 2 there are acoustic coupling paths, labeled 26-1, and 26-2, between the loudspeaker 22-1 and the microphone 21-1, and between the loudspeaker 22-2 and the microphone 21-2, but the echo cancelers 23-1 and 23-2 act to eliminate the signal coupled from the loudspeaker into the microphone.
The inventor has presented adaptive digital filters suitable for use in such an echo canceler in Japanese patent application Nos. 164770/1986 and 163677/1986 and a corresponding U.S. patent application Ser. No. 070,773, filed July 7, 1987. FIG. 3 shows the first of these adaptive digital filters. This adaptive digital filter comprises M basic sections connected in series (where M is a positive integer). Basic sections 1 through M-1 each comprise a second-order recursive digital filter and a second-order nonrecursive digital filter. Basic section M, the final section, comprises a second-order recursive digital filter. Adjacent basic sections are connected by scalers that multiply by Q.sub.m and by 1/Q.sub.m (m=1 to M). Each basic section also has a first-order nonrecursive digital filter comprising a scaler R.sub.m for multiplying the first-order delayed output u.sub.m (k-1) by the coefficient R.sub.m, an adder 31 for adding the result to the 0th-order delayed output u.sub.m (k), a scaler S.sub.m for multiplying the output of the adder 31.sub.m by the coefficient S.sub. m, a variable-coefficient scaler c.sub.m (k) for multiplying the result u.sub.m (k) by a variable coefficient c.sub.m (k), a variable-coefficient scaler d.sub.m (k) for multiplying the first-order delayed output u.sub.m (k-1) by a variable coefficient d.sub.m (k), and an adder 32.sub.m for adding the outputs of the two variable-coefficient scalers c.sub.m (k) and d.sub.m (k). The outputs from the adders 32.sub.m are added one after the other and the output from the adder 33.sub.M is the output y(k) of the filter with respect to the input x(k).
The coefficients Q.sub.m of the scalers Q.sub.m (m=1 to M) are chosen so that the mean square values of the first-order delayed outputs u.sub.m (m-1) (m=1 to M) are equal to the mean square value of the input signal x(k) (a white-noise signal). The coefficients S.sub.m of the scalers S.sub.m (m=1 to M) are chosen so that the means square values of their outputs u.sub.m (k) (m=1 to M) are equal to the mean square value of the input signal x(k). The coefficients R.sub.m of the scalers R.sub.m (m=1 to M) are chosen so that when x(k) is a white-noise signal, the signals u.sub.m (k-1) (m=1 to M) are orthogonal. These selections increase the rate of convergence of the variable coefficients c.sub.m (k) and d.sub.m (k) (also called the adaptive parameters) of the variable-coefficient scalers.
The adaptive digital filter shown in FIG. 3 displays excellent convergence characteristics when the input signal x(k) is a white-noise signal. When the input signal has nonwhite characteristics, as in a voice signal, however, u.sub.m (k-1) and u.sub.m (k) are not orthogonal, and their mean square values are unequal, so the convergence characteristics of the variable coefficients (adaptive parameters) c.sub.m (k) and d.sub.m (k) are degraded.